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| 01:48.26 | compu_85 | hi |
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| 02:30.02 | vader- | Hello |
| 02:31.49 | vader- | Do you guys know if there is a SIP client for android that will allow me to add multiple extensions to a phone for inbound and outbound calling? |
| 02:32.25 | Samot | Depends on the SIP client. |
| 02:32.59 | Samot | X-Lite, Zoiper, Bria; they will all offer a different amount of SIP accounts. |
| 02:33.08 | vader- | 3CX? |
| 02:33.29 | Samot | 3CX is designed for 3CX but you can use it with other PBXes. |
| 02:33.40 | Samot | How many SIP accounts do you need? |
| 02:34.19 | vader- | i have a couple businesses that i need to answer different phone numbers for and need to be able to dial out from |
| 02:34.30 | vader- | i was looking at something like grasshopper but i rather roll my own |
| 02:35.58 | Samot | Well I'm not sure how many are supported in the Android/iOS platforms. |
| 02:36.18 | Samot | The more SIP accounts, the more traffic and resources.. |
| 02:37.32 | vader- | ya |
| 02:37.51 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:41d2:5895:dc75:2fde) |
| 02:39.55 | Samot | And it would be best advised to do the signaling over TCP instead of UDP. |
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| 03:43.15 | igcewieling1 | Samot: CSIPSimple supports a LOT of registrations. |
| 03:44.00 | Samot | There you go. |
| 03:44.32 | Samot | on Android? |
| 03:44.54 | igcewieling1 | Yes. |
| 03:45.21 | Samot | vader- ^^^ |
| 03:47.01 | igcewieling1 | It also supports TCP and in theory TLS, DTLS, STRP. IIRC, I had to load certificates into the android OS and never got it working well over the flaky 3G connection I was using. |
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| 03:56.14 | igcewieling1 | IIRC Polycom VVX are nice. I have a VVX600 (touch screen, which I hate) with 14 registrations. |
| 03:57.39 | igcewieling1 | My main phone is a VVVX 410 which has 12 line buttons for registrations |
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| 05:03.21 | DoYouNo | what's the current gold standard for free sip clients? |
| 05:03.27 | DoYouNo | on windows |
| 05:03.30 | DoYouNo | err |
| 05:03.31 | DoYouNo | linux |
| 05:03.32 | DoYouNo | sorry |
| 05:03.41 | DoYouNo | I forget I'm running linux sometimes :) |
| 05:16.06 | vader- | i need a decent desk phone as well |
| 05:16.18 | vader- | but i need to be able to direct everything to my cell phone too |
| 05:18.18 | drmessano | DoYouNo: No Such thing.. it's whatever works for you |
| 05:19.26 | drmessano | Zoiper and Blink are good |
| 05:19.50 | drmessano | IMHO, Bria is the gold standard for mobile. |
| 05:20.02 | drmessano | They've just dropped the ball with Linux |
| 05:53.48 | igcewieling1 | The gold standard for SIP clients are Polycom VVX hardphones. 8-|. Better to ask what the least bad clients are. |
| 06:01.55 | drmessano | Except he asked about Linux |
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| 06:16.58 | DoYouNo | I'm trying to connect asterisk to google voice and am getting this error when I dial a number: == Everyone is busy/congested at this time (1:0/0/1) |
| 06:17.22 | DoYouNo | it's connected via xmpp fine |
| 06:27.04 | DoYouNo | Unable to add Google ICE candidates as ICE support not available or no candidates available |
| 06:29.41 | drmessano | You need to enable ICE |
| 06:30.17 | DoYouNo | I have icesupport=yes |
| 06:30.38 | drmessano | Where? |
| 06:30.49 | DoYouNo | in rtp.conf |
| 06:32.30 | drmessano | Under the [general] section? |
| 06:32.38 | DoYouNo | yes |
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| 06:33.01 | drmessano | Did you reload after adding it? |
| 06:33.24 | drmessano | Pastebin your rtp.conf |
| 06:34.04 | DoYouNo | http://pastebin.com/z2ndzVgk |
| 06:34.14 | Rasputin3711 | What is a good guide for self-prepare for entry level - dcaa certification? |
| 06:37.28 | DoYouNo | [Jan 11 00:37:15] ERROR[32729][C-00000003]: chan_motif.c:987 jingle_add_google_candidates_to_transport: Unable to add Google ICE candidates as ICE support not available or no candidates available |
| 06:37.32 | DoYouNo | that's the full error |
| 06:40.32 | DoYouNo | did they change icesupport=yes to icesupport=true? |
| 06:40.36 | DoYouNo | or do they mean the same thing? |
| 06:41.27 | drmessano | Remove it |
| 06:41.35 | drmessano | It defaults to enabled |
| 06:43.47 | DoYouNo | doesn't help |
| 06:46.12 | drmessano | What are you using as your guide to set this up? |
| 06:46.23 | DoYouNo | https://ubuntuforums.org/showthread.php?t=2259082 |
| 06:49.57 | drmessano | Did you build Asterisk from source? |
| 06:50.00 | DoYouNo | yes |
| 06:50.19 | drmessano | Where did you get pjproject? |
| 06:50.19 | DoYouNo | I'm pretty sure it loaded the modules good |
| 06:50.26 | DoYouNo | from the repos |
| 06:51.08 | drmessano | you grabbed the dev packages from the repos? |
| 06:51.36 | DoYouNo | I see this: [Jan 11 00:45:52] WARNING[3085]: res_hep_rtcp.c:153 load_module: res_hep is not loaded or running; declining module load |
| 06:52.00 | drmessano | you grabbed the dev packages from the repos? |
| 06:52.06 | DoYouNo | sudo apt-get install libpjsip2 |
| 06:52.15 | DoYouNo | I need the dev package? |
| 06:52.17 | drmessano | What version of Ubuntu? |
| 06:52.33 | DoYouNo | it's an older version |
| 06:52.37 | DoYouNo | armbian |
| 06:52.41 | DoYouNo | for orange pi |
| 06:52.57 | drmessano | You need a recent version of PjProject |
| 06:53.07 | drmessano | What version of Asterisk? |
| 06:55.12 | DoYouNo | Asterisk 14.2.1 |
| 06:55.35 | DoYouNo | I've got the bleeding edge |
| 06:56.07 | drmessano | You need to remove the pjproject package and try a much newer one |
| 06:56.18 | DoYouNo | ok |
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| 09:27.56 | defswork | We're passing calls inward calls back out via a dedicated sip trunk to a third party system and they do some routing based on the CLI - they are looking at the SIP From: header - expecting the cli to be here - <sip:xxxxxx@10.11.12.13> where the xxxxxx are - is this the right place to get the CLI from ? |
| 09:29.01 | defswork | because we our from header contains the trunk username - From: "xxxxxx" <sip:supertrunk@77.107.121.160> and the CLI in xxxxxx |
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| 10:44.23 | [sID] | I did webrtc and it works on Firefox browser, but the strange thing is that on another compiling anymore. |
| 10:44.42 | [sID] | gets an error tcptls_stream_close: SSL_shutdown() failed: 1 |
| 10:44.53 | [sID] | What could this mean? |
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| 11:01.42 | blinky_ | hi all, looking for some help identifying a problem. We have 2 asterisk boxes on different sites, each has a trunk out to the voip supplier and all external call are working fine. On each site the calls between the extension on that box work fine. We then have a trunk between the two boxes so we can make extension calls between the boxes, this is where the issue is. When a call is made from site 1 to site 2, or vise versa |
| 11:01.42 | blinky_ | the call starts fine and then degrades badly. I presume it is dropping packets but I am unable to figure out what is causing it. Any pointers on how to solve this? |
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| 11:20.33 | BarthezZ | I'm trying to prevent Asterisk from propagating some information, but I isn't doing what I want... this is on 1.8.23 using chan_sip; Phone-A dials to PBX-B which dials to Phone-C; When dialing Phone-C it indicates a 100 trying and 180 Ringing, which get propagated to Phone-A; Unfortunately Phone-A can't handle the ringing properly, is there a way to prevent asterisk from propagating the ringing; the dia |
| 11:20.39 | BarthezZ | lplan basically just is Dial(SIP/Phone-C) |
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| 11:43.48 | file | defswork: "right" is relative - it can be in the From, Remote-Party-ID, or P-Asserted-Identity header |
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| 13:59.41 | dan_j | Hi. Has anyone experienced mixmonitor missing the first few seconds of each call? |
| 13:59.52 | dan_j | I can't find it in the bug tracker. |
| 14:01.59 | dan_j | Mixmonitor is started prior to the call being answered so it shouldnt be missing off the first few seconds, but it is. |
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| 14:02.55 | saxpga | Hello to everyone |
| 14:03.52 | saxpga | I have a issue with my server asterisk |
| 14:04.09 | saxpga | i installed all things |
| 14:04.19 | saxpga | ans i can make and receive a call |
| 14:04.43 | saxpga | but i failed to configure my voicemail with ODBC |
| 14:05.39 | saxpga | Can someone help me to resove my issue?? please |
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| 14:36.36 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:38.03 | defswork | file, hmm - ok - not right then but most common ? where does * get it from on incoming calls ? |
| 14:38.45 | file | I wouldn't say most common, and if you don't set fromuser in the configuraton then we won't change it |
| 14:38.51 | file | and it is configurable where it gets it |
| 14:39.27 | file | (at least in PJSIP land, chan_sip I believe is fixed at PAI, RPID, and then From) |
| 14:40.02 | file | actually it's fixed for both it seems - PAI, fallback to RPID, fallback to From |
| 14:41.02 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
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| 14:45.25 | [TK]D-Fender | blinky_, You still haven't told (or far better shown) us the problem yet. |
| 14:49.30 | dan_j | blinky_: There are some routers (usually supplied by ISPs) which have issues and cause voip quality to degrade even though tests show no packet loss. In the UK, the EE routers do that. |
| 14:49.58 | dan_j | At least they used to. I dont know if they have fixed the issue. |
| 14:50.27 | [TK]D-Fender | missed a line in there |
| 14:50.51 | [TK]D-Fender | quality degrade = networking issue |
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| 14:51.50 | dan_j | [TK]D-Fender: With the EE routers, it sounds to the ear like packet loss, but any other type of data transfer is totally fine with no packet loss. |
| 14:52.07 | [TK]D-Fender | because TCP retransmits |
| 14:52.12 | [TK]D-Fender | You can't compare to other protocols |
| 14:52.34 | [TK]D-Fender | TCP has to give up to actually count something as a "loss" |
| 14:53.00 | [TK]D-Fender | As for quality, jitter can be just as bad as certain amounts of loss |
| 14:53.09 | dan_j | Even UDP is fine. Just not SIP Audio packets. |
| 14:53.58 | [TK]D-Fender | SIP audio = UDP |
| 14:54.16 | Samot | RTP always = UDP |
| 14:54.34 | [TK]D-Fender | Other UDP protocols implement application layer retransmits |
| 14:54.48 | [TK]D-Fender | So just saying "other UDP" doesn't mean the test is fair |
| 14:55.15 | [TK]D-Fender | RTP = no app-layer retransmit |
| 14:55.15 | dan_j | Doesnt matter what you say. All EE Brightbox routers caused poor audio quality. And swapping the router out always fixed it. |
| 14:55.32 | dan_j | Test sending UDP (without retransmits) all pass. |
| 14:55.43 | dan_j | It is a known issue. Not something I'm imagining. |
| 14:55.47 | [TK]D-Fender | Sure it does. Matter as far as mainting a proper testing methodology |
| 14:56.12 | [TK]D-Fender | And we don't know the actual scenario or if he's using what you say you swapped on your side |
| 14:56.33 | [TK]D-Fender | it worked for YOU and may not even be what he's using, let alone actuallyy work for HIM even if he did. |
| 14:56.41 | dan_j | True, it's just an idea for him to think about if he cant see any actual packet loss. |
| 14:57.17 | [TK]D-Fender | This is "correlation vs causation" with the lowest sample size possible so far |
| 14:57.45 | Samot | Do those routers have SIP ALG or SPI Firewall active on them? |
| 14:57.54 | dan_j | my sample size is fairly large as it has been reported by other SIP providers. |
| 14:58.14 | dan_j | I dont think the SIP ALG settings are accessible on those routers. I can't remember actually. |
| 14:58.22 | [TK]D-Fender | Samot, Neither would hit quality issues |
| 14:58.30 | dan_j | Correct |
| 14:58.35 | [TK]D-Fender | Samot, RTP quality of delivery, not SIP |
| 14:58.47 | Samot | Just checking. |
| 14:58.56 | [TK]D-Fender | CAFFEINE :p |
| 14:59.12 | blinky_ | Sorry guys, been a long day. We are trying to get the company to invest in better hardware as we are running a BT Business Hub at the moment that has no controls at all. on top of that we are running the Asterisk on a VM that is also running other servers and think this is all contributing to the issue. I have been testing today and have cleaned the line a lot at the moment but I do not hold any hope for going forward. |
| 14:59.35 | dan_j | blinky_: what router are you using at the moment? |
| 15:00.03 | dan_j | [TK]D-Fender: Most links here are from people reporting issues. https://www.google.co.uk/search?q=ee+sip+packet+loss&ie=utf-8&oe=utf-8&client=firefox-b&gfe_rd=cr&ei=kEZ2WMXUCKLS8AeY35vAAQ#q=ee+brightbox+router+voip+quality |
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| 15:00.11 | blinky_ | The fiber router that BT has supplied. Not good |
| 15:01.26 | [TK]D-Fender | VM under load is a serious suspect |
| 15:02.12 | [TK]D-Fender | If you have a spare host I'd recommend migrating to that just to remove the other usage as a suspect. |
| 15:04.06 | dan_j | blinky_: Which one? The BT Home Hub has SIP ALG issues which can't be disabled. But that shouldn't affect call quality. |
| 15:05.07 | blinky_ | The issue is I migrated from the previous host as it was failing, we have not been give permission to buy additional hardware yet. I have just found an option within the BT Business Hub of SIP ALG. On reading it is suggested that it be disabled, is it something that you would suggest for better quality and stability? |
| 15:05.47 | [TK]D-Fender | good to have off regardless |
| 15:06.00 | dan_j | I find that with a Business Hub and SIP ALG Enabled, I can't even register. Once disabled, I can register and dial without any issue. So I cannot comment on whether the quality will improve. |
| 15:07.01 | dan_j | However, I have clients using the BT Business Hub without any problems. Which Business Hub is it? |
| 15:07.27 | dan_j | 3,4,5 shouldn't have any issues. The older ones did. |
| 15:07.42 | blinky_ | I will have a look at disabling it when people leave the network. I am putting forward requests for a new SonicWall TZ300 as we need to VPN access and better security solution for this site. |
| 15:08.12 | blinky_ | Hub 5 |
| 15:08.48 | dan_j | Hub 5 with SIP ALG disabled works fine. If you still have problems, it's probably not the router. |
| 15:13.21 | *** join/#asterisk kharwell (kharwell@nat/digium/x-irlubzqwsaixbtzt) |
| 15:13.21 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:20.17 | *** join/#asterisk _corrupt (~chalumnin@184.75.223.195) |
| 15:21.47 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-cmrhqwbusqpoizdf) |
| 15:21.47 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 15:22.39 | BarthezZ | Don't like to spam, but still having this issue: I'm trying to prevent Asterisk from propagating some information, but I isn't doing what I want... this is on 1.8.23 using chan_sip; Phone-A dials to PBX-B which dials to Phone-C; When dialing Phone-C it indicates a 100 trying and 180 Ringing, which get propagated to Phone-A; Unfortunately Phone-A can't handle the ringing properly, is there a way to preve |
| 15:22.45 | BarthezZ | nt asterisk from propagating the ringing; the dialplan basically just is Dial(SIP/Phone-C) |
| 15:22.47 | blinky_ | What software would you suggest as a complete package for SIP and IAX? We are running Elastix2.5 at the moment but the boss want me to look into updating to a current system. |
| 15:27.11 | dan_j | blinky_: Not really the right channel for that type of question as the answer (if given) may be bias (such as FreePBX which is asterisk based). And the answer is largely based on your requirements which we dont know. |
| 15:27.52 | blinky_ | ok fair point, I am looking at straight asterisk at the moment, will have a look at frePBX as well, cheers |
| 15:28.31 | dan_j | FreePBX is basically just a GUI which runs asterisk in the background and deals with the dialplan for you. Don't expect to be able to play around with the dialplan if you use something like FreePBX. |
| 15:29.23 | Samot | That's not entirely true. You can play around all you want, you just need to understand the pre-existing dialplan. |
| 15:34.33 | dan_j | Ok, no entirely true, but still requires more research than simply editing a dialplan you've made yourself. |
| 15:34.45 | dan_j | Not* |
| 15:34.50 | Samot | In some cases. |
| 15:35.06 | Samot | I have a FreePBX box that doesn't touch a lot of the predefined code. |
| 15:36.10 | Samot | In regards to dialplan. |
| 15:37.01 | *** join/#asterisk Phrohdoh (4222c1a6@gateway/web/cgi-irc/kiwiirc.com/ip.66.34.193.166) |
| 15:44.11 | Phrohdoh | Hi all, asterisk is reporting that it is calling my number and playing a sound file but my mobile phone never actually rings. What am I doing wrong? https://gist.github.com/Phrohdoh/adc07b59a5d21e3fecd214d0b142d186 |
| 15:47.39 | *** join/#asterisk slack1 (~slack@103.254.153.99) |
| 15:50.54 | [TK]D-Fender | Phrohdoh, Look at the actual SIP debug of your call |
| 15:55.56 | Samot | Are you getting the voicemail? |
| 15:56.07 | Phrohdoh | I am not. |
| 15:56.09 | Samot | But a sip debug would be very helpful. |
| 15:56.27 | Phrohdoh | [TK]D-Fender: Ok, I will do some reading about that and try to get back to you asap. |
| 15:56.41 | Samot | sip set debug on |
| 15:56.46 | Samot | make the call |
| 15:56.53 | Phrohdoh | oh, that's easy |
| 15:56.54 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 15:59.33 | Phrohdoh | https://gist.github.com/Phrohdoh/adc07b59a5d21e3fecd214d0b142d186#file-sip-debug-md |
| 16:00.49 | [TK]D-Fender | Reliably Transmitting (NAT) to 166.78.105.67:5060: |
| 16:00.55 | [TK]D-Fender | no sane provider is behind NAT |
| 16:01.06 | Samot | Yup. |
| 16:01.43 | [TK]D-Fender | Contact: <sip:user_9muj8x8f@104.43.142.33:5060> <- also please confirm that is your WAN IP |
| 16:02.06 | [TK]D-Fender | and that you have 5060, and your rtp.conf specified ports (all UDP) forwarded / open to your server |
| 16:08.04 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 16:12.25 | Phrohdoh | sys admin says that isn't the trunk's WAN IP (nor is it the office's) |
| 16:13.37 | Samot | Sys Admin will show the PBX IP |
| 16:13.42 | Phrohdoh | He said that is an IP in MS' Azure, no idea why it would be connecting to that. |
| 16:13.44 | Samot | Not the WAN IP if you are behind NAT. |
| 16:13.54 | Samot | Asterisk SIP Settings |
| 16:14.22 | Phrohdoh | Is https://gist.github.com/Phrohdoh/adc07b59a5d21e3fecd214d0b142d186#file-sip-conf helpful at all? |
| 16:14.59 | Samot | No, I just told you were to look. |
| 16:15.01 | *** join/#asterisk Hornet-Wing (~Hornet-Wi@80.4.184.162) |
| 16:15.04 | Samot | Asterisk SIP Settings. |
| 16:15.30 | Phrohdoh | I don't have access to a PBX GUI. |
| 16:15.44 | Samot | asterisk -rx "sip show settings" |
| 16:16.02 | Phrohdoh | Got it, thanks |
| 16:17.02 | Phrohdoh | https://gist.github.com/Phrohdoh/adc07b59a5d21e3fecd214d0b142d186#file-asterisk-sip-settings |
| 16:17.26 | [TK]D-Fender | <PROTECTED> |
| 16:17.26 | [TK]D-Fender | Externaddr: 104.43.142.33:0 |
| 16:17.35 | [TK]D-Fender | You gave it a hostt, and that's how it is resolving |
| 16:17.47 | Samot | ^ Yuppers. |
| 16:17.56 | [TK]D-Fender | Either using the wrong hostname, or the wrong address is assigned tto it |
| 16:18.23 | Phrohdoh | Ok, thank you! I will send that to the sysadmin and try to get that resolved. |
| 16:18.41 | Samot | That doesn't get set in Sys Admin |
| 16:18.46 | Samot | Asterisk SIP Settings |
| 16:19.51 | *** join/#asterisk krzee (~k@openvpn/community/support/krzee) |
| 16:19.54 | Phrohdoh | Are you saying the system administrator can't find the right information and set that in asterisk? |
| 16:20.15 | [TK]D-Fender | mixup of terms |
| 16:20.19 | Samot | Yes. |
| 16:20.25 | Samot | Sys Admin is a module in FreePBX |
| 16:20.39 | Phrohdoh | Oh, sorry for the confusion. |
| 16:21.07 | [TK]D-Fender | Technically your wording did imply the person, not the module |
| 16:21.20 | Samot | Yes, it did. |
| 16:21.22 | [TK]D-Fender | So Samot this one's on you ;) |
| 16:21.27 | Samot | Yup. |
| 16:21.52 | [TK]D-Fender | Lowest entry on the SNAFU scale naturally |
| 16:25.07 | *** join/#asterisk sekil (~sekil@cable-89-216-222-10.dynamic.sbb.rs) |
| 16:25.09 | TandyUK | so, what UK audio packs are there for asterisk which are actually complete? |
| 16:25.33 | TandyUK | i see loads of packs offering like 600 or so prmopts, but having been throug hand transcribed every message theres more like 1200 in total |
| 16:25.51 | TandyUK | if i were to get a new one done, in a british male voice, would there be a market for it |
| 16:25.55 | Phrohdoh | How do I reload the sip config in the asterisk cli? |
| 16:26.02 | TandyUK | noting that it would be an actual complete pack, not just half of it! |
| 16:26.27 | TandyUK | reload modules iirc |
| 16:26.31 | Chainsaw | TandyUK: I would prefer a female UK voice to replace my current incomplete pack. |
| 16:26.41 | Phrohdoh | `sip reload`, easy enough |
| 16:26.45 | TandyUK | Chainsaw: that would be the second step after getting the uk male pack done |
| 16:26.48 | [TK]D-Fender | Phrohdoh, "sip reload" |
| 16:26.58 | TandyUK | im looking at the male pack costing about £2k to get done |
| 16:27.08 | TandyUK | havent found a female voice artist yet |
| 16:27.27 | TandyUK | they key is finding someone who is goign to be available for years to come for additional custom prompts people might want |
| 16:27.53 | TandyUK | and assuming it was a female, how much would yo ube prepared to spend on a complete pack of sounds? |
| 16:28.03 | Phrohdoh | TandyUK: Audio synthesis? |
| 16:28.18 | TandyUK | hells no, actual humans wil ldo all my recordings |
| 16:28.29 | Phrohdoh | heh, ok |
| 16:28.33 | Chainsaw | TandyUK: A soft highland accent is amongst the easiest to understand. |
| 16:28.38 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 16:28.43 | Chainsaw | TandyUK: Something like say... Cereproc Heather. |
| 16:29.17 | Chainsaw | TandyUK: (It's what I use with my Sygic navigation software in the car) |
| 16:29.36 | TandyUK | that even soudns shit though |
| 16:29.50 | TandyUK | i can tell from 'her' first sentence that its a computer generated voice |
| 16:30.03 | Chainsaw | TandyUK: Sure, but as an accent choice... |
| 16:30.14 | TandyUK | yeah, certainly avoid brummie lol |
| 16:31.19 | Chainsaw | TandyUK: And for a TTS voice that one is the only one to pronounce "Goldhay Way" correctly. Any other TTS engine, including Google turn it into Goldie Way or Gold-HAY way. |
| 16:32.39 | Phrohdoh | https://gist.github.com/Phrohdoh/0218316690ea6273bf93de0fb5e7136d |
| 16:33.40 | *** join/#asterisk bof22 (~Thunderbi@185.13.183.107) |
| 16:33.43 | *** join/#asterisk saul (~hubert@165.98.98.114) |
| 16:33.54 | Samot | SIP/2.0 480 Temporarily Unavailable |
| 16:34.01 | Samot | That's coming back from the provider. |
| 16:35.29 | Samot | [Jan 11 10:33:35] NOTICE[25751]: pbx_spool.c:413 attempt_thread: Call failed to go through, reason (5) Remote end is Busy |
| 16:36.06 | Phrohdoh | Yeah I don't know what exactly is busy in that case. My phone isn't in a call or anything. |
| 16:36.27 | Phrohdoh | Well, technically I am placing two calls, but my phone doesn't ring. |
| 16:37.13 | Samot | Your provider is returning that. |
| 16:37.26 | Samot | Can you dial your cell without issue from a phone? |
| 16:37.37 | Samot | On the PBX? |
| 16:37.59 | *** join/#asterisk craigify (~craigify@162.216.46.168) |
| 16:38.12 | Phrohdoh | I can test that via originate on the cli, correct? |
| 16:38.45 | *** join/#asterisk Milos (~Milos@pdpc/supporter/student/milos) |
| 16:38.57 | craigify | has anyone else experienced thread locks with chan_sip and mysql realtime, eventually hosing chan_sip. I've seen a recent bug about this. I'm wondering if switching to odbc is more stable. |
| 16:39.13 | craigify | 13.x |
| 16:39.20 | Samot | You can but that's what is having the issue now.. |
| 16:39.26 | *** join/#asterisk Y04NN (~y04nn@178.18.54.206) |
| 16:39.28 | Samot | I'm trying to verify that a normal outbound call works. |
| 16:39.33 | Samot | To where you are trying to dial. |
| 16:39.47 | Samot | I want to see if the provider is going to return the same error. |
| 16:39.48 | Phrohdoh | Sorry, I'm too ignorant when it comes to asterisk to know how to do that then. |
| 16:39.56 | Samot | You don't have a phone registered? |
| 16:40.43 | Phrohdoh | We do not, apparently. |
| 16:41.35 | Phrohdoh | (per the sysadmin human) |
| 16:43.26 | Samot | Here's the thing.. |
| 16:43.31 | Samot | You are making the call just fine. |
| 16:43.43 | Samot | Your provider is returning the 480 error. |
| 16:43.45 | Phrohdoh | So `originate` calls my cell just fine but doesn't play `hello-world` |
| 16:43.59 | Samot | You said you phone doesn't ring. |
| 16:44.15 | Samot | I just showed you a log entry that says the call was busy |
| 16:44.17 | Phrohdoh | I just now tried to make the call via `originate` |
| 16:44.21 | Samot | OK. |
| 16:44.32 | Samot | So you have a new debug to show? |
| 16:44.47 | Phrohdoh | I do, I'll gist it. |
| 16:45.32 | Phrohdoh | https://gist.github.com/Phrohdoh/0218316690ea6273bf93de0fb5e7136d#file-asterisk-help-md |
| 16:47.17 | Samot | Reliably Transmitting (NAT) to 166.78.105.67:5060: <-- Please fix this. |
| 16:47.26 | Samot | This was the first issue pointed out. |
| 16:47.42 | Samot | Your provider is not behind NAT. |
| 16:47.50 | Samot | Or at least they shouldn't be. |
| 16:48.20 | igcewieling1 | that message is almost always harmless |
| 16:49.09 | Phrohdoh | I don't know what to fix here nor does the sysadmin. I'll bug another person in the office that may know. |
| 16:49.40 | Samot | nat=no |
| 16:49.43 | Samot | in the trunk settings. |
| 16:49.56 | Samot | "almost" |
| 16:50.37 | igcewieling1 | from sip.conf.sample: For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE ; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the ; other, then valid peers with settings differing from those in the general section will ; be discoverable. |
| 16:51.08 | Phrohdoh | <--- Transmitting (NAT) to 166.78.105.67:5060 ---> |
| 16:51.29 | Phrohdoh | Er ignore that, I forgot to `sip reload` |
| 16:51.57 | Phrohdoh | Ok so the log _still_ contains: Reliably Transmitting (NAT) to 166.78.105.67:5060: |
| 16:52.15 | Samot | Did the last call go through? |
| 16:52.19 | Samot | Did you hear you cell ring? |
| 16:53.28 | Phrohdoh | The most recent `originate` call did go through, my cell rang, I answered it and the call was silent for ~5 seconds then hungup. |
| 16:53.44 | Samot | Did you say anything?! |
| 16:53.45 | Phrohdoh | I then placed a call via a .call file and that did not ring my cell |
| 16:53.55 | igcewieling1 | are you sure directmedia=no is set? |
| 16:54.00 | Phrohdoh | Why would I say anything, it is waiting for silence. |
| 16:54.02 | Samot | You realize you are running AMD? |
| 16:54.20 | Samot | Which is a Human vs. Machine detection. |
| 16:54.24 | Phrohdoh | Ah, duh, ok sorry |
| 16:54.41 | Samot | If you don't say anything, it's going to die if the "silence" period is exceeded. |
| 16:55.39 | Phrohdoh | This time I spoke and it still hungup on me |
| 16:55.44 | Phrohdoh | (without playing any sound) |
| 16:56.19 | [TK]D-Fender | <igcewieling1> that message is almost always harmless <- often. Or if a provider uses separate CC vs media servers you're DOA |
| 17:02.20 | Samot | We need to see a full debug of a call that hangs up on you. |
| 17:02.28 | Samot | asterisk -rvvvvvvvvv |
| 17:02.30 | Samot | sip set debug on |
| 17:03.31 | saul | is there a way to disable Call Forwarding All on all but certain extensions? I've some real genius users that enable it by accident and then people start to complain that they get calls going to other departments :( |
| 17:03.54 | saul | i'd rather not disable it completely because IT uses forwarding to their cell phones when they're not at their desks |
| 17:04.52 | Samot | saul: Depends on where it is set. |
| 17:04.57 | Samot | Some phones do it locally. |
| 17:05.04 | Samot | Meaning you add the CF number on the phone.. |
| 17:05.09 | Samot | Or are they dialing a feature code? |
| 17:09.45 | Phrohdoh | https://gist.github.com/Phrohdoh/0218316690ea6273bf93de0fb5e7136d |
| 17:10.32 | Phrohdoh | Since this is via `kazoo` will the dialplan default to kazoo-outgoing? |
| 17:11.59 | [TK]D-Fender | Reliably Transmitting (NAT) to 166.78.105.67:5060: |
| 17:12.01 | [TK]D-Fender | Still wrong |
| 17:12.07 | [TK]D-Fender | nat=NO |
| 17:12.08 | [TK]D-Fender | ^ |
| 17:12.11 | saul | Samot: i think people are using the softkey :( |
| 17:12.12 | Phrohdoh | It is set to no. |
| 17:12.26 | saul | Samot: i suspect then asterisk is powerless to stop it since it's happening on the phone ? |
| 17:12.34 | [TK]D-Fender | You did not apply changes then |
| 17:12.47 | saul | i will disable the feature code anyway |
| 17:12.53 | [TK]D-Fender | Contact: <sip:user_9muj8x8f@166.78.105.67:5060> |
| 17:12.53 | Phrohdoh | I have `sip reload`ed twice since that change was written to disk |
| 17:13.13 | [TK]D-Fender | The IP has changed, can you confirm that this is correctt? |
| 17:13.36 | [TK]D-Fender | Show us the configs masking only the secret |
| 17:17.10 | Phrohdoh | https://gist.github.com/Phrohdoh/0218316690ea6273bf93de0fb5e7136d#file-sip-conf |
| 17:19.18 | Phrohdoh | Is `register => user_9muj8x8f:8k2qucm26zed@d79abb.s.zswitch.net:5060` in sip.conf valid? |
| 17:24.44 | [TK]D-Fender | not relevant |
| 17:24.57 | Phrohdoh | ok |
| 17:25.11 | [TK]D-Fender | https://gist.github.com/Phrohdoh/0218316690ea6273bf93de0fb5e7136d#file-sip-conf |
| 17:25.15 | [TK]D-Fender | Line 34 <---- |
| 17:25.23 | Phrohdoh | ugh |
| 17:25.24 | [TK]D-Fender | what partt of "nat=NO" was I unclear on? |
| 17:25.24 | Phrohdoh | sorry |
| 17:25.31 | Phrohdoh | [TK]D-Fender: chill out dude |
| 17:25.45 | [TK]D-Fender | your register also has to come after everything else under [general] |
| 17:25.49 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:26.00 | Phrohdoh | alright, fixed that too |
| 17:26.08 | [TK]D-Fender | "canreinvite" is deprecated. exchange for "directmedia" |
| 17:26.21 | Phrohdoh | done |
| 17:26.28 | [TK]D-Fender | externhost=d79abb.s.zswitch.net |
| 17:26.34 | [TK]D-Fender | this is also looking pretty wrong. |
| 17:26.41 | [TK]D-Fender | that is the host you are CALLING in your peer |
| 17:26.49 | Phrohdoh | Changing that is what changed the IP |
| 17:26.50 | [TK]D-Fender | 17 vs 24 |
| 17:27.10 | [TK]D-Fender | externhost is supposed to be YOUR hostname |
| 17:27.20 | [TK]D-Fender | Hos is it that is tthe hostt you ahve fo kazoo? |
| 17:28.00 | [TK]D-Fender | How is it that this the host you have for kazoo? |
| 17:28.26 | Phrohdoh | I don't know, that is what the sysadmin told me to use for the kazoo hostname. |
| 17:29.41 | [TK]D-Fender | externhost is YOUR host |
| 17:29.52 | [TK]D-Fender | You can'tt go shoving the provider address there |
| 17:30.15 | Phrohdoh | Yet when I had it set to acaexpress.com you said the ip was wrong |
| 17:30.31 | [TK]D-Fender | No, you said it was wrong after I asked if it was right |
| 17:31.08 | [TK]D-Fender | 17 = YOUR hostname or ip address |
| 17:31.22 | [TK]D-Fender | 24 = your PROVIDER's hostname or ip address |
| 17:31.29 | [TK]D-Fender | just get these right and confirmed |
| 17:31.51 | [TK]D-Fender | Then we'll try a new call |
| 17:32.06 | [TK]D-Fender | If it fails show tthe newly updatted config and the attempt |
| 17:32.17 | Phrohdoh | Will do |
| 17:35.27 | [TK]D-Fender | BRB |
| 17:38.15 | *** join/#asterisk sekil (~sekil@cable-89-216-222-10.dynamic.sbb.rs) |
| 17:41.02 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 18:01.08 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 18:01.08 | *** mode/#asterisk [+o DivideBy0] by ChanServ |
| 18:02.41 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 18:06.17 | *** join/#asterisk sekil (~sekil@cable-89-216-222-10.dynamic.sbb.rs) |
| 18:11.03 | *** join/#asterisk defsdoor (~andy@cpc35-sutt4-2-0-cust184.19-1.cable.virginm.net) |
| 18:20.18 | *** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:c8b7:18de:ca08:cc3) |
| 18:25.15 | *** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com) |
| 18:39.24 | *** join/#asterisk Hornet-Wing (~Hornet-Wi@80.4.184.162) |
| 19:00.37 | *** join/#asterisk BakaKuna (~BakaKuna@145.129.205.133) |
| 19:04.07 | Phrohdoh | So there are some sounds (in /var/lib/asterisk/sounds) that have both .wav and .g722 or .gsm formats. Can I specify which to play with the Playback command? The docs I've read say to _not_ include file extension when doing `Playback(hello-world)` for example. |
| 19:04.56 | Samot | It's takes the best option for the call. |
| 19:04.57 | yang | [TK]D-Fender: Which is the best audio codec to be used along with Polycom Soundpoint phones if bandwith is not an isssue ? |
| 19:05.00 | Samot | Transcodes as needed. |
| 19:05.35 | Phrohdoh | Well that isn't working as desired in my case. A coworker's phone plays the sound just fine while mine is just a high-pitched tone. |
| 19:06.57 | Samot | What codecs are you using? |
| 19:07.04 | *** join/#asterisk ThomasKeller (~Thomas@vmx.ethz.ch) |
| 19:07.34 | Phrohdoh | Not sure, where could I find that in a config? |
| 19:07.51 | Samot | Is this straight Asterisk or FreePBX? |
| 19:07.57 | Phrohdoh | asterisk |
| 19:08.06 | Samot | So you're not running FreePBX? |
| 19:08.14 | Phrohdoh | Correct |
| 19:08.27 | Samot | Then it would be in your SIP settings. |
| 19:08.37 | Samot | In in the general context or for the peer. |
| 19:09.43 | Samot | I just looked at the scroll back and you're using ulaw. |
| 19:10.13 | Samot | [general] |
| 19:10.13 | Samot | context=internal |
| 19:10.33 | Phrohdoh | > 9 allow=ulaw |
| 19:10.46 | Phrohdoh | So should I be using something else? |
| 19:10.53 | Samot | But you shouldn't have your general context as the same as your context for internal devices. |
| 19:11.28 | Phrohdoh | I didn't set that up (outside of modifying nat etc), what should be removed from the general context? |
| 19:11.50 | Samot | You "general" context should really blackhole people. |
| 19:12.20 | Samot | Incoming calls to the PBX from the provider(s) or device(s) should be sent to their own context. |
| 19:12.39 | Phrohdoh | I don't expect any incoming calls to the pbx, fwiw |
| 19:12.46 | Samot | Doesn't matter. |
| 19:12.50 | Phrohdoh | How do I 'blackhole' someone? |
| 19:12.52 | Samot | If you get compromised.... |
| 19:13.01 | Phrohdoh | true |
| 19:13.07 | Samot | They're calls will be processed as "internal" calls. |
| 19:13.46 | Samot | [from-external-notallowed] |
| 19:14.00 | Samot | exten => s,1,HangUp() |
| 19:14.03 | Samot | Or something like that. |
| 19:14.20 | Phrohdoh | And then [general] context=from-external-notallowed? |
| 19:14.29 | Samot | Or whatever you name it. |
| 19:14.31 | Samot | But yes. |
| 19:14.51 | Phrohdoh | Ok, thanks I'll change that. What should I do regarding file playback? |
| 19:15.02 | Samot | Show the two different calls. |
| 19:15.06 | Samot | asterisk -rvvvvvvvvv |
| 19:15.10 | Samot | sip set debug on |
| 19:27.34 | Phrohdoh | https://gist.github.com/Phrohdoh/5f52c2f999633717f83da6666f749efe |
| 19:34.53 | Samot | Rename the .gsm file to something else. |
| 19:35.01 | Samot | Or move it somewhere else. |
| 19:35.14 | Samot | Make it so that .wav is the "best" (low rate) option. |
| 19:36.43 | Phrohdoh | Ok, so one thing I'm a bit confused on is that my /var/lib/asterisk/sounds/ does not contain a `tt-monkeys.*` at all |
| 19:37.53 | Samot | <PROTECTED> |
| 19:38.00 | Samot | In /var/lib/asterisk/sounds/en |
| 19:38.10 | Phrohdoh | Ahh ok thanks |
| 19:38.30 | Samot | Your default language is English, so that's where the English files are. |
| 19:40.49 | *** join/#asterisk seiggy (~seiggy@74.203.105.194) |
| 19:42.39 | Phrohdoh | I think I am doing something incorrectly. |
| 19:42.40 | Phrohdoh | > Looking for s in from-external-not-allowed (domain 104.43.142.33) |
| 19:43.22 | *** join/#asterisk tzafrir (~tzafrir@37.26.147.203) |
| 19:50.05 | Phrohdoh | I don't know why it'd be running in that context: https://gist.github.com/Phrohdoh/90f1fc35679bcacaf0934b19ad5482f9 |
| 19:52.01 | Phrohdoh | Eh disregard that I suppose it still rang my phone |
| 19:52.11 | Phrohdoh | But the wav file I played this time was also just a buzz (not the expected sound) |
| 19:52.51 | Samot | Is this file in the proper format? |
| 19:53.00 | Samot | The wav format as to be correct. |
| 19:53.28 | Phrohdoh | I do not know for sure but it is a wav that asterisk already had (I didn't record it), if that is worth anything |
| 19:53.54 | Samot | You need a wider test bed. |
| 19:54.08 | Samot | You have two calls, one plays the file without issue and the other does not. |
| 19:57.49 | Phrohdoh | I've done this with two files now, not that two is a lot, but it isn't one. This is frustrating. |
| 19:58.20 | Samot | You said you called two different phones. Yours and someone else's. |
| 19:58.36 | Samot | Your phone gets a weird buzz, they hear the audio fine. |
| 19:58.51 | Phrohdoh | Yes, I have reproduced this with two separate sound files. |
| 19:59.07 | Samot | So now you need more people to call. |
| 20:00.02 | Samot | If you make 10 calls and 9 other calls work fine but your cell phone keeps hearing buzzing.. |
| 20:00.19 | Samot | Then it's a pretty limited focus on where the issue is. |
| 20:01.34 | Samot | I suggest that you get a device registered that makes actual calls so you can figure out if this is a playback issue or something more. |
| 20:08.07 | Phrohdoh | I called another coworker and he didn't hear the sound either https://gist.github.com/Phrohdoh/59347dc4c7ee6aba7117a95b29ab5304 |
| 20:08.22 | Phrohdoh | I'll talk with my boss and the sysadmin about registering a device. |
| 20:15.19 | *** join/#asterisk Hornet-Wing (~Hornet-Wi@cpc22-new25-2-0-cust520.5-1.cable.virginm.net) |
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| 21:17.15 | zargonovski | Hi all |
| 21:19.10 | zargonovski | Hi |
| 21:20.30 | zargonovski | need help with hylafax please |
| 21:20.32 | zargonovski | anyone |
| 21:23.54 | zargonovski | need help , any experts aroud ? |
| 21:24.01 | zargonovski | around * |
| 21:27.28 | Phrohdoh | Just try asking your question |
| 21:27.35 | Phrohdoh | If someone knows they'll hopefully answer |
| 21:30.23 | zargonovski | I couldn't configure the hylafax server with elastix PBX |
| 21:31.13 | zargonovski | especially the IAX part |
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| 23:53.25 | snadge | asterisk 13.8-cert2 .. registered to another asterisk server.. when calling out, sends invite, receives 401 unauth (normal), sends ACK |
| 23:53.47 | snadge | then instead of sending another invite, with the username and nonce.. it just says in the log.. failed to authenticate on invite |
| 23:55.48 | snadge | sip trace isn't helpful.. trying to think of what could cause it to that |